Grandstream HT801 SIP ATA (1x WAN, 1x FXS)

R835.00 inc. VAT

The Grandstream HT801 is a compact single-port FXS analogue telephone adapter (ATA) that connects one existing analogue phone or fax machine to a VoIP network. Supporting 1 SIP profile through a single FXS port and a single 10/100 Mbps Ethernet port, it delivers HD voice quality, T.38 Fax-over-IP, TLS/SRTP security, and automated provisioning in Grandstream’s most affordable and smallest ATA — ideal for single-line deployments, individual remote workers, or large-scale commercial rollouts.

In stock

Add to Wishlist
Add to Wishlist

Description

The Simplest Path from Analogue Phone to VoIP

The HT801 is a single port analogue telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution for residential and office environments. Its ultra-compact size, voice quality, advanced VoIP functionality, security protection, and auto provisioning options enable users to take advantage of VoIP on analogue phones. Built on Grandstream’s market-leading SIP ATA/gateway technology with millions of units successfully deployed worldwide, the HT801 is the most streamlined and cost-effective way to bring a single analogue handset or fax machine onto a modern SIP-based VoIP platform.

Key Highlights

  • 1× RJ11 FXS port for connecting a single analogue phone or fax machine
  • 1 SIP profile supported via a single 10/100 Mbps auto-sensing Ethernet port
  • Ultra-compact desktop form factor — minimal desk footprint
  • Ideal for both individual use and large-scale commercial IP voice deployments

HD Voice Quality & Broad Codec Support

The HT801 delivers exceptional voice quality across a wide range of application environments, with wideband codec support ensuring compatibility with virtually any VoIP platform or SIP trunk provider. Dynamic jitter buffer and advanced line echo cancellation ensure consistently clear audio regardless of network conditions.

  • G.711 (Annex I/II), G.723.1, G.729A/B, G.726, iLBC, and Opus codec support with dynamic jitter buffer and advanced line echo cancellation
  • T.38 compliant Fax-over-IP up to 14.4 kbps with automatic G.711 fax pass-through fallback
  • Caller ID: Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID
  • Layer 2 (802.1Q VLAN) and Layer 3 (ToS, DiffServ, MPLS) QoS support for prioritised voice traffic

Full Telephony Feature Set

Telephony features include caller ID display and block, call waiting, flash, blind and attended transfer, call forward, hold, do not disturb, and 3-way conferencing — the complete feature set business users expect, delivered through an analogue handset they already have.

  • Caller ID, call waiting, call transfer, call forward, hold, and DND
  • 3-way voice conferencing
  • Message waiting indication and multi-language prompts
  • Flexible dial plans for call routing control
  • Failover SIP server automatically switches to a secondary server if the primary loses connection

Security, Provisioning & UCM Integration

TLS and SRTP security encryption protect calls and accounts, with a unique security certificate per unit providing device-level protection out of the box. Automated provisioning options include TR-069 and XML config files for streamlined mass deployment across large sites. Zero Configuration provisioning is available when used with Grandstream’s UCM series of IP PBXs, and full GDMS compatibility allows centralised management alongside other Grandstream devices at no additional cost.

  • AES encryption with unique per-unit security certificate
  • TLS and SRTP for secure call transmission
  • TR-069 and XML config file provisioning for automated mass deployment
  • Zero Configuration provisioning with Grandstream UCM IP PBX series
  • GDMS compatible for free centralised cloud management
  • Web UI, SSH, and Telnet for local management

 

Proudly sold by the Ringwood Digital Store.

Additional information

Weight 0.5 kg

Reviews

There are no reviews yet.

Be the first to review “Grandstream HT801 SIP ATA (1x WAN, 1x FXS)”

Your email address will not be published. Required fields are marked *